Media/WebRTC/ReleaseNotes/56

From MozillaWiki
Jump to: navigation, search

Firefox 56 WebRTC/WebAudio Release Notes:

Full listing of all WebRTC & WebAudio bugs marked as Fixed in Firefox 56:

WebRTC bugs: Bugzilla search for WebRTC related bugs marked Fixed in Firefox 56

WebAudio bugs: Bugzilla search for WebAudio bugs marked Fixed in Firefox 56

Noteworthy Changes:

bug 1341285 Update WebRTC code to webrtc.org stable branch 57

Bug tickets fixed in Firefox 56 that affect WebRTC or Web Audio (full list):

Audio/Video:Cubeb :

bug 1385481 Update cubeb from upstream to 42da8d2

bug 1384053 Update cubeb from upstream to 09aeb59

bug 1381015 Update cubeb from upstream to a329c6a

bug 1380233 Update cubeb from upstream to bb18984

bug 1376000 All Tabs Crash in libpulse

bug 1375873 all tabs crash in nightly (thread '<unnamed>' panicked at 'SinkInfo contains invalid flags')

bug 1374494 Update cubeb-pulse-rs to to 6451581

bug 1374275 Update cubeb from upstream to 18153b9

bug 1373475 cubeb-pulse-rs: Panic in pulse_format_to_cubeb_format

bug 1373213 Repair cubeb logging

bug 1197045 Add some basic audio hardware/driver/format information to about:support

Audio/Video:GMP (Gecko Media Plugin):

bug 1385115 dom/media/gmp/GMPChild.cpp:438:33: error: 'XUL_LIB_FILE' was not declared in this scope

bug 1383580 EME video playback fails due to hitting CDM IPC shmem limit

bug 1382883 [EME] Pass Widevine Verified Media Pipeline sigs to Widevine CDM.

bug 1381720 [EME] Update content_decryption_module.h

bug 1376957 Widevine CDM 970 fails to load on Windows 7

bug 1375708 Netflix broken on Linux in Firefox 54

bug 1374499 Increase number of shmems pre-allocated by PChromiumCDM.

bug 1372080 [EME] Reorder frames coming out of Widevine CDM

Audio/Video:MediaStreamGraph (MSG):

bug 1378067 Label more runnables in dom/media (GraphDriver.cpp, CubebUtils.cpp, DOMMediaStream.cpp)

bug 1372247 Make the AudioOutputObserver a member of the MediaEngineWebRTCMicrophoneSource so it's not a global singleton anymore

bug 1371719 Don't create an MSG if it's not going to be used immediately

bug 1341555 Label runnables in dom/media/MediaStreamGraph.cpp

bug 1330360 Use multiple MediaStreamGraph in the same process

bug 1312623 Intermittent dom/media/tests/mochitest/test_peerConnection_setLocalAnswerInStable.html | application crashed [@ mozilla::CycleCollectedJSContext::ProcessMetastableStateQueue]

Audio/Video:Media Recording:

bug 1382718 MediaRecorder throws RecordErrorEvent on valid window mediaStream with no further info

WebAudio:

bug 1382897 Intermittent dom/media/webaudio/test/test_nodeCreationDocumentGone.html | application crashed [@ mozilla::OffTheBooksMutex::AssertCurrentThreadOwns] after IsAcquired() && mOwningThread == PR_GetCurrentThread(), at BlockingResourceBase.cpp:400

bug 1375235 AudioChannelService::IsWindowActive() creates unnecessary AudioChannelWindow objects

Core (General) WebRTC:

bug 1384874 Build fail on OSX 10.11.6 after Bug 1368030 landed

bug 1384826 Media: WebRTC: Fix build config for MIPS

bug 1384655 Build with --enable-alsa --enable-tests is broken by webrtc::AudioDeviceLinuxALSA

bug 1383137 Intermittent /webrtc/RTCPeerConnection-generateCertificate.html | generateCertificate() with 0 expires parameter should generate expired cert - assert_less_than_equal: expected a number less than or equal to 1500650276906 but got 1500650276907

bug 1383069 Intermittent TEST-UNEXPECTED-PASS | /webrtc/RTCPeerConnection-setLocalDescription.html | setLocalDescription(offer) should never resolve if connection is closed in parallel - expected FAIL

bug 1382972 Intermittent /webrtc/RTCPeerConnection-addIceCandidate.html | Add valid candidate should never resolve when pc is closed - Peer connection is closed

bug 1382681 Crash in java.lang.RuntimeException: Camera is already stopped! at org.webrtc.videoengine.VideoCaptureAndroid.stopCaptureOnCameraThread(VideoCaptureAndroid.java)

bug 1380430 Backport current webrtc.org support for RtpStreamId (and the much better handling of variable size RtpHeaderExtensions)

bug 1379982 mach webrtc-gtest does not work on mac

bug 1379836 Fix AEC Logging

bug 1378412 Build error on Linux 32 bit due to a warning in task_queue_libevent.cc with clang 4.0

bug 1377959 jvm_android.cc passes va_list to varags methods

bug 1376357 Enable ESLint for dom/media/*.js

bug 1375238 WebRTC 57 fails to build with Clang i386 due to -Wc++11-narrowing

bug 1374465 Webrtc video inbound-rtp does not include framesDecoded

bug 1373858 Intermittent dom/media/tests/mochitest/test_peerConnection_stats.html | candidate-pair.bytesSent was tested.

bug 1373015 Permafailing dom/media/tests/mochitest/test_peerConnection_stats.html | candidate-pair.state is succeeded. value=cancelled

bug 1372687 [Mdm2] Intermittent test_peerConnection_basicVideoRemoteHwEncoder.html | java-exception org.mozilla.gecko.mozglue.GeckoLoader$AbortException: abort() called from :0xdd6216d9 () at org.mozilla.gecko.mozglue.GeckoLoader.abort(GeckoLoader.java:532)

bug 1372509 Self-XSS XUL Injection in about:webrtc

bug 1371000 Firefox not seeing all windows available to share

bug 1343515 RTPStats inbound-rtp.jitter is wrong when isRemote=true

bug 1341285 Update WebRTC code to webrtc.org stable branch 57

WebRTC:Audio/Video:

bug 1382095 Mainthread-refcounts on MediaEngineSource in setLastPrefs runnable can sidestep proper cleanup on shutdown.

bug 1379743 Screensharing with resize constraints is broken (graphics garbage) (regression)

bug 1379392 Firefox crashes (UAF) whenever camera is not readable (NotReadableError) in OSX. regression.

bug 1377093 Build mediapipeline tests on BSDs

bug 1377078 ooura_fft.cc/ooura_fft_sse2.cc has '//' in #include patch which causes build failure

bug 1376885 Implement a way to deny gUM requests in continuous integration

bug 1374938 mediaDevices.ondevicechange and cam contraints are broken on Mac OSX

bug 1374640 Multiple calls to getUsermedia return the same window / screen

bug 1374204 [Firefox for Android] set preference for hardware encoder to true in nightly

bug 1368030 Intermittent dom/media/tests/mochitest/test_getUserMedia_basicScreenshare.html | application terminated with exit code 5

bug 1365852 Intermittent browser/base/content/test/webrtc/browser_devices_get_user_media_unprompted_access_queue_request.js | leaked 1 window(s) until shutdown [url = chrome://browser/content/webrtcIndicator.xul]

bug 1364038 getUserMedia returns an unusable stream after removing and reconnecting camera

bug 1355048 Add a WebrtcVideoDecoder MediaDataDecoder class.

bug 1328169 mediaconduit_unittests are no longer being built (or run)

bug 1213414 Implement channelCount audio constraint

WebRTC:Networking:

bug 1383575 Remove TURNS log warning

bug 1383272 Fix RTP parsing error for RtpStreamId

bug 1373103 nicer log filled with read callback messages if TCP is used

bug 1368159 Tweak error log line to warn or info at ice_ctx.c:710

bug 1343755 Label runnables in netwerk/sctp/datachannel/

bug 1339906 Update IceCandidatePairStats

WebRTC:Signaling:

bug 1383020 mediaconduit_unittests.cpp: suggest explicit braces to avoid ambiguous 'else' [-Werror=dangling-else]

bug 1379400 Intermittent dom/media/tests/mochitest/test_peerConnection_stats.html | candidate-pair.bytesReceived is a sane number (20,000<>800,000) for a short test. value=801806

bug 1377803 Remove unnecessary plarena.h #includes

bug 1374440 max-message-size should only be emitted with EOR support

bug 1373450 MaxMessageSize zero is ambiguous

bug 1373144 Ignore multiple MSID's as preparation for Transceivers

bug 1371841 PeerConnectionImpl::RecordEndOfCallTelemetry sends telemetry when no connection information was exchanged

bug 1371161 Port SDP file parser to LibFuzzer

bug 1370601 Make it possible for offerer and answerer to switch roles in jsep_session_unittest.cpp

bug 1355947 Use TestNrSocket to build a fake ICE implementation for testing

bug 1305813 Ensure RID isn't empty when constructing RTP header

bug 1264479 Implement RTCPeerConnection attributes: currentLocalDescription pendingLocalDescription currentRemoteDescription pendingRemoteDescription

bug 1196974 Remove mozDontOfferDataChannel/mozBundleOnly from RTCOfferOptions